1. Field of Invention
The present invention generally relates to a method and hardware structure for audio signal decoding, and more particularly, to an inverse-modified discrete cosine transform and overlap-add method and hardware structure for MPEG Layer3 audio signal decoding.
2. Description of Related Art
Digital audio signal processing is widely used. This is because the digital audio signal immunity to noise is higher than the analog signal. However, since it is quite often demanded to process a large amount of data within a very short time and still needs to maintain the effect of high audio quality, a lot of the audio signal compression standards have been developed. The motion picture experts group (abbreviated as MPEG) standard is widely accepted due to its high compression rate and low distortion. MPEG, using the different sensitivity of the human ear to different frequency bands, assigns fewer bits to the audio to which the human ear is not so sensitive, to achieve the objective of compression.
Furthermore, in order to accommodate different levels of audio quality with the compression method, MPEG is further divided into Layer1, Layer2 and Layer3. Generally speaking, the higher the level of the layer, the more complicated the compression method, the distortion of the corresponding recovered audio signal is much less, and the effect is better.
The encoding process of MPEG can be divided into the encoder and the decoder portions. In the encoder portion, the audio data is processed and converted into 32 data sub-bands by using the analysis sub-band filter bank. Then, the data belonging to different bands can be assigned to different bits according to the psycho-acoustical model that simulates the artificial ear acoustic effect. Afterwards, the objective of the compression can be achieved via quantization. Finally, the data is sent out in a specific data format framing.
The decoder portion looks like the reverse operation of the encoder. The data is unpacked first, and after the inverse quantization process, the 32 data sub-bands are integrated into the original audio data by using the synthesis sub-band filter bank.
As to the MPEG-II audio encoding standard, multi-channel audio encoding is further provided, while all the other aspects are basically the same as the MPEG I. Multi-channel audio can be divided into the Left (L) and Right (R) channel audio transmitted via the basic transmission channels T0, T1, and the Central (C), Left Surround (LS) and Right Surround (RS) channel audio transmitted via the extended transmission channels T2, T3, T4. The multichannel decoder is needed for the MPEG-II audio decoding to reconstruct the multichannel audio signal.
The MPEG LAYER3 compression standard, using the MPEG Layer3 (MP3) compression algorithm, is widely used in the application of digital broadcast and multimedia. As to the digital audio signal compression, MP3 is the most complicated algorithm, providing the highest compression rate within MPEG. MP3 utilizes the inverse-modified discrete cosine transform (hereinafter abbreviated as IMDCT) and the sub-band coding techniques, whereby MP3 can achieve such high compression rate.
The hardware structure of MPEG Layer1and Layer2 decoders has already been physically implemented by many researchers. However, there is no appropriate hardware structure to implement MP3. Most of the hardware structure design nowadays is implemented using the general digital signal processor (abbreviated as DSP). This design 5 utilizes program control to achieve the objective. However, a large amount of memory is needed for this design to store the program code, and thus the hardware burden and area is increased, so that the performance of the entire system cannot achieve the optimum.